Telecommunications networks currently rely to a large extent upon the Signalling System no. 7 (SS7) as the mechanism for controlling call connections and for handling the transfer of signalling information between signalling points of the networks. Typically, one or more application and user parts at a given signalling point will make use of SS7 to communicate with peer application and user parts at some other signalling point. Examples of user parts are ISUP (ISDN User Part) and TUP (Telephony User Part) whilst examples of application parts are INAP (Intelligent Network Application Part) and MAP (Mobile Application Part). The conventional SS7 protocol stack includes Message Transfer Parts MTP1, MTP2, and MTP3 which handle the formatting of signalling messages for transport over the physical layer, error correction and detection, as well as various routing functions.
SS7 typically uses the same physical transport layer as is used for transporting actual user information, e.g. voice and facsimile information. In Europe, the conventional physical transport mechanism is a time division multiplexed Synchronous Transport Mechanism (STM) known as E.1. In the US, a similar transport mechanism known as T.1 is used.
There has been considerable interest of late amongst the telecommunications community in using non-standard (i.e. non-conventional within the telecommunications industry) “bearer” transport mechanisms in telecommunications networks to carry user data, for example, voice traffic. The reasons for this are related both to improvements in efficiency as well as potential cost savings. Much consideration has been given for example to the use of Internet Protocol (IP) networks to transport user information between network nodes. IP networks have the advantage that they make efficient use of transmission resources by using packet switching and are relatively low in cost due to the widespread use of the technology (as opposed to specialised telecommunication technology). There is also interest in using other transport mechanisms such as ATM.
The standard ISUP which deals with the setting-up and control of call connections in a telecommunications network is closely linked to the standard bearer transport mechanism and does not readily lend itself to use with other non-standard transport technologies such as IP and ATM. As such, several standardisation bodies including the ITU-T, ETSI, and ANSI, are currently considering the specification of a protocol for the control of calls, which is independent of the underlying transport mechanism. This can be viewed as separating out from the protocol, Bearer Control functions which relate merely to establishing the parameters (including the start and end points) of the “pipe” via which user plane data is transported between nodes, and which are specific to the bearer transport mechanism. The new protocol, referred to as Bearer Independent Call Control (BICC), retains Call Control functions such as the services invoked for a call between given calling and called parties (e.g. call forwarding), and the overall routing of user plane data. FIG. 1a illustrates the conventional integrated Call Control and Bearer Control structure of ISUP whilst FIG. 1b illustrates the proposed new separated structure. In place of BICC, alternative protocols such as SIP may be used.
It is noted that at the junctions between different bearer networks, i.e. between different transport media, a gateway is present which requires both the CC functions and BC functions. The splitting of the CC and BC control derives in part from the Gateway Decomposition work which was carried out by the IETF SS7IP, SIGTRAN and MEGACO working groups, ETSI Tiphon Project, ITU SG16 Study Group 16, ATM Forum and Multiswitching Service Forum (MSF) on establishing an architecture and requirements for decomposed gateways.
As a result of the CC/BC split, a new interface is exposed between the CC functions and BC functions. A Gateway Control Protocol is required to enable coupling between the CC functions and BC functions when a node is implemented in a separated environment. The term for this standardised interface protocol is the ‘Media Gateway Control Protocol’ (MGCP). This protocol is being developed by the ITU Study Group 16 (H.GCP) and in the IETF MEGACO (MGCP) working group. In ITU Study Group 16 and IETF MEGACO, the CC function is know as ‘Media Gateway Controller (MGC)’ and the BC function is known as ‘Media Gateway (MG)’. The need for the MGCP is illustrated in FIG. 2, which illustrates two peer gateway nodes which communicate with one another at both the CC level and the BC level. It will be appreciated that the definition of a MGCP will allow a Media Gateway Controller to be used with any type of Media Controller (and vice versa) as long as both utilise the MGCP.